The public switched telephone network (PSTN) has evolved into an efficient real-time, multi-media communication session tool wherein users can pick up any one of nearly one billion telephones and dial any one of nearly one billion endpoints. Several developments have enabled this automated network, such as numbering plans, distributed electronic switching and routing, and networked signaling systems.
Unfortunately, the PSTN is not currently capable of routing an actual communication session on anything other than an address that conforms to the hierarchy present in the PSTN since telephone numbers and their parts are used to discover a path to an endpoint of the communication. Portable mechanisms require a phantom or shadow number to direct the communication session through the network.
Similar to the manner in which the PSTN is based on a hierarchy, the Internet is based on an Internet Protocol (IP). IP messages are routed or forwarded from one link te the next (i. e., from a source of the data flow to a destination of the data flow). Each IP packet contains an IP address, which, in Internet Protocol version 4 (IPv4), has 32 bits. Each IP address also has a certain number of bits dedicated to a network portion and a certain number of bits dedicated to a host portion.
IP routers are used to take a packet from one network (or link) and place it onto another network (or link). Tables are located within IP routers that contain information or criteria used to determine a best way to route a packet. An example of this information may be the state of network links and programmed distance indications. Unfortunately, IP routers typically route packets by destination IP address, which does not assist in finding a proper route for transportation. There are some exceptions to this routing system, however, by using intelligent devices on both sides of a network domain, it is possible to allocate a temporary address to route a packet through a network and restore the original address on the far side of the network when the packet leaves the network. This is the basis for many current virtual private network (VPN) products and is understood in the art.
Another exception to the routing system includes multi-protocol label. switching (MPLS). MPLS is based on a technology developed by Cisco Systems, Inc. of San Jose, Calif. called tag switching. This method of routing IP packets allows a destination IP address to potentially be separated from the route that the packet actually takes through a network. One of the best uses of MPLS is to create a VPN or virtual leased lines (VLL). The MPLS tags can effectively encapsulate the routing of data packets through a network.
In summary, it is concluded that data networks base real forwarding of IP packets on IP destinations. IP destinations, in turn, are associated with network topology and, like the telephone network, are used to deliver packets. MPLS tags and paths can provide override forwarding for IP packets based on a set of rules that is tied to the IP address portion used for routing, such as, for example, a forward equivalence class (FEC).
To ensure that the network elements (e.g., switches in the telephone network, routers in the data network) can perform their associated tasks, they should know the status of adjacent communication links and available routes; signaling systems are used to provide this information. In telephone networks, typical signaling systems used are either SS7 compliant or are equivalent to SS7. The signaling system provides information about individual links, link sets, routes, etc. In data networks, protocols such as border gateway protocol (BGP), interior gateway protocol (IGP), open shortest path first (OSPF), etc., are used to determine link states and routes.
In the telephone networks, the signaling system is also used to establish an end-to-end path (i.e., ISDN User Part (ISUP)) through the network. Unfortunately, in IP networks, there is no end-to-end path allocation. Instead, to engage in a communication session, a system to associate endpoints with names or purposes is needed.
There are currently no known universal registries on the Internet. A universal registry with the domain name E164.com has been proposed by NetNumber.com, Inc., of Lowell, Mass. This universal registry development is based on a proposal by NueStar, Inc., which is now responsible for administering the North American numbering plan (NANP). This proposal calls for using the current domain name service (DNS) and formatting the numbers into URLs in a way that can be resolved using DNS servers. In this manner, each telephone number could be registered into a DNS server and distributed to all other DNS servers. The tail end of a DNS query could be a resource record, which points to a lightweight directory access protocol (LDAP) directory server.
The suggestion from the ITU to use Universal Portable Telephone (UPT) numbers for IP endpoints to avoid overlapping traditional wired telephone numbers is valid and would allow for addressable IP endpoints. It is possible to combine the above two proposals to enable Internet calling to and from the PSTN. Unfortunately, there are several limitations to this technology. These limitations include: DNS distribution and replication has significant latency; DNS address resolution can be slow; DNS servers may not be capable of handling the number of projected addresses; DNS servers are incapable of managing duplicate entries (except through round robin techniques); DNS employs parallel update mechanisms, which may result in unintentional duplicate entries; private network addresses or addressing gateways may result in duplicate entries or matches; no policy exists to handle the management of the resources requested; and, no solution exists to handle the number overlap between the PSTN and the data networks.
Due to most current telecommunication endpoints receiving service through a PSTN-based system, a gateway is used to facilitate a multi-media flow between a packet data network and a PSTN. Gateways are installed at edges between data networks and voice networks, wherein the gateways are used to convert multi-media (and signaling) to ensure communication. There are several strategies for routing calls received by gateways to other gateways described in the art. Two of these strategies are full mesh routing and hierarchical routing. Full mesh routing is the standard method described in most of the softswitching architectures. Session initiation protocol (SIP) is the inter-softswitch signaling system because it supports an anywhere-to-anywhere signaling model. In this model, all softswitches have a virtual connection to all other softswitches for completing calls. Routing tables are utilized that can be used to direct traffic to a softswitch based on policy provided by the softswitch maker.
Unfortunately, when running a network that comprises many softswitches, the owner of the network has many different points of policy management to maintain a full mesh. Such policy management issues include assuring that each softswitch knows the IP address of each other softswitch and what telephone numbers or PSTN to which they connect. When running softswitches from multiple vendors, further management issues arise. The management issues are then more complicated due to the fact that the equipment may be managed through different links.
When the number of softswitches deployed grows large, the sharing of different routes is likely. In the full mesh routing arrangement, the routing of calls may be difficult since several different egress softswitches may be full or not functioning. For example, if a carrier has thirty softswitches that can handle national long distance, and the network is running at about 50% full, then each originating softswitch will likely have to try an average of 15 separate softswitches before finding one with a non-blocked route. This search effort can be greatly reduced if a pure random distribution is implemented; however, it is assumed that some routes would be preferred over others due to cost or quality, thereby exacerbating the problem.
Certain simple gateways, such as, but not limited to, the Cisco AS5300, can forward SIP-based call requests to a SIP proxy server. Unfortunately, these gateways have low densities and frequently lack the sophistication of softswitches in setting up routing policies. These routers, therefore, cannot be interconnected to create networks without a softswitch controller.
Therefore, guiding real-time packet flows through certain thresholds, which is generally required to create a high-quality border between various IP networks, is important. Without proper guidance, the packets would flow whichever way the networks would allow, thereby subjecting packets to disruptive paths, as well as upstream and downstream failures.